Added A/V sync tests with drifting clocks.

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
@@ skipping 53 matching lines @@
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
   void DestroyCalls();
 
   void CreateSendConfig(size_t num_video_streams,
                         size_t num_audio_streams,
                         Transport* send_transport);
   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
 
+  void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock, float speed);
   void CreateFrameGeneratorCapturer();
   void CreateFakeAudioDevices();
 
   void CreateVideoStreams();
   void CreateAudioStreams();
   void Start();
   void Stop();
   void DestroyStreams();
 
   Clock* const clock_;
@@ skipping 101 matching lines @@
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_